ADAPTIVE AUDIO CONSTRUCTION
    21.
    发明申请

    公开(公告)号:US20190281404A1

    公开(公告)日:2019-09-12

    申请号:US16424409

    申请日:2019-05-28

    Abstract: Described herein is a method for creating an object-based audio signal from an audio input, the audio input including one or more audio channels that are recorded to collectively define an audio scene. The one or more audio channels are captured from a respective one or more spatially separated microphones disposed in a stable spatial configuration. The method includes the steps of: a) receiving the audio input; b) performing spatial analysis on the one or more audio channels to identify one or more audio objects within the audio scene; c) determining contextual information relating to the one or more audio objects; d) defining respective audio streams including audio data relating to at least one of the identified one or more audio objects; and e) outputting an object-based audio signal including the audio streams and the contextual information.

    Method for improving perceptual continuity in a spatial teleconferencing system

    公开(公告)号:US09628630B2

    公开(公告)日:2017-04-18

    申请号:US14430841

    申请日:2013-09-25

    CPC classification number: H04M3/561 H04M3/569 H04M2203/5072

    Abstract: The present document relates to audio conference systems. In particular, the present document relates to improving the perceptual continuity within an audio conference system. According to an aspect, a method for multiplexing first and second continuous input audio signals is described, to yield a multiplexed output audio signal which is to be rendered to a listener. The first and second input audio signals (123) are indicative of sounds captured by a first and a second endpoint (120, 170), respectively. The method comprises determining a talk activity (201, 202) in the first and second input audio signals (123), respectively; and determining the multiplexed output audio signal based on the first and/or second input audio signals (123) and subject to one or more multiplexing conditions. The one or more multiplexing conditions comprise: at a time instant, when there is talk activity (201) in the first input audio signal (123), determining the multiplexed output audio signal at least based on the first input audio signal (123); at a time instant, when there is talk activity (202) in the second input audio signal (123), determining the multiplexed output audio signal at least based on the second input audio signal (123); and at a silence time instant, when there is no talk activity (201, 202) in the first and in the second input audio signals (123), determining the multiplexed output audio signal based on only one of the first and second input audio signals (123).

    Spectral and Spatial Modification of Noise Captured During Teleconferencing
    24.
    发明申请
    Spectral and Spatial Modification of Noise Captured During Teleconferencing 审中-公开
    在电话会议期间捕获的噪声的光谱和空间修改

    公开(公告)号:US20140278380A1

    公开(公告)日:2014-09-18

    申请号:US14192649

    申请日:2014-02-27

    CPC classification number: G10L21/0232 G10L21/0208 H04M3/18 H04M3/568 H04M9/082

    Abstract: In some embodiments, a method for modifying noise captured at endpoints of a teleconferencing system, including steps of capturing noise at each endpoint, and modifying the captured noise to generate modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set. In other embodiments, a teleconferencing method including steps of: at endpoints of a teleconferencing system, determining audio frames indicative of audio captured at each endpoint, each of a subset of the frames indicative of noise but not a significant level of speech; and at each endpoint, generating modified frames indicative of modified noise having a frequency-amplitude spectrum which matches a target spectrum and a spatial property set which matches a target spatial property set, and generating encoded audio including by encoding the modified frames. Other aspects are systems configured to perform any embodiment of the method.

    Abstract translation: 在一些实施例中,一种用于修改在电话会议系统的端点处捕获的噪声的方法,包括在每个端点处捕获噪声的步骤,以及修改所捕获的噪声,以产生具有与目标频谱匹配的频率幅度谱和空间属性的修正噪声 设置与目标空间属性集匹配。 在其他实施例中,电话会议方法包括以下步骤:在电话会议系统的端点处,确定指示在每个端点处捕获的音频的音频帧,指示噪声而不是显着级别的语音的帧的子集中的每一个; 并且在每个端点处,生成指示具有与目标频谱匹配的频率幅度谱和与目标空间属性集匹配的空间属性集合的频率幅度谱的修改后的帧,以及通过编码修改的帧来生成编码音频。 其他方面是被配置为执行该方法的任何实施例的系统。

    Layered Mixing for Sound Field Conferencing System
    25.
    发明申请
    Layered Mixing for Sound Field Conferencing System 有权
    声场会议系统分层混音

    公开(公告)号:US20140240447A1

    公开(公告)日:2014-08-28

    申请号:US14166065

    申请日:2014-01-28

    CPC classification number: H04N7/152 G10L19/008 G10L25/78 H04M3/567 H04M3/569

    Abstract: A conferencing server (100) receives incoming bitstreams (I1, I2, I3, I4, I5) carrying media data from respective conferencing endpoints (110, 120, 130, 140, 150); receives a mixing strategy (M) specifying properties of at least one outgoing bitstream (O1, O2, O3, O4, O5) and requiring at least one additive media mixing step; and supplies at least one outgoing bitstream by executing, in a processor (103) and a memory (102) with a plurality of memory spaces, a run list of operations selected from a predefined collection of primitives and realizing the received mixing strategy. A pre-processor (104) in the server derives said run list repeatedly and dynamically while taking into consideration determined momentary activity in each incoming bitstream. In embodiments, the run list may be derived by (a) pruning of an initial run list, (b) constrained or non-constrained minimization of a cost function, or (c) automatic code generation.

    Abstract translation: 会议服务器(100)从相应的会议端点(110,120,130,140,​​150)接收携带媒体数据的传入比特流(I1,I2,I3,I4,I5); 接收指定至少一个传出比特流(O1,O2,O3,O4,O5)的属性并需要至少一个添加介质混合步骤的混合策略(M) 并且通过在具有多个存储器空间的处理器(103)和存储器(102)中执行从预定义的图元集合中选择的操作的运行列表并实现所接收的混合策略来提供至少一个输出比特流。 服务器中的预处理器(104)在考虑每个输入比特流中确定的瞬时活动的同时,重复地和动态地导出所述运行列表。 在实施例中,可以通过(a)修剪初始运行列表,(b)成本函数的约束或非约束最小化,或(c)自动代码生成来导出运行列表。

    Post-teleconference playback using non-destructive audio transport

    公开(公告)号:US11115541B2

    公开(公告)日:2021-09-07

    申请号:US16691487

    申请日:2019-11-21

    Abstract: Teleconference audio data including a plurality of individual uplink data packet streams, may be received during a teleconference. Each uplink data packet stream may corresponding to a telephone endpoint used by one or more teleconference participants. The teleconference audio data may be analyzed to determine a plurality of suppressive gain coefficients, which may be applied to first instances of the teleconference audio data during the teleconference, to produce first gain-suppressed audio data provided to the telephone endpoints during the teleconference. Second instances of the teleconference audio data, as well as gain coefficient data corresponding to the plurality of suppressive gain coefficients, may be sent to a memory system as individual uplink data packet streams. The second instances of the teleconference audio data may be less gain-suppressed than the first gain-suppressed audio data.

    Method of rendering one or more captured audio soundfields to a listener

    公开(公告)号:US11089421B2

    公开(公告)日:2021-08-10

    申请号:US16908568

    申请日:2020-06-22

    Abstract: A computer implemented system for rendering captured audio soundfields to a listener comprises apparatus to deliver the audio soundfields to the listener. The delivery apparatus delivers the audio soundfields to the listener with first and second audio elements perceived by the listener as emanating from first and second virtual source locations, respectively, and with the first audio element and/or the second audio element delivered to the listener from a third virtual source location. The first virtual source location and the second virtual source location are perceived by the listener as being located to the front of the listener, and the third virtual source location is located to the rear or the side of the listener.

    Normalization of soundfield orientations based on auditory scene analysis

    公开(公告)号:US10708436B2

    公开(公告)日:2020-07-07

    申请号:US15956470

    申请日:2018-04-18

    Abstract: Embodiments are described for a soundfield system that receives a transmitting soundfield, wherein the transmitting soundfield includes a sound source at a location in the transmitting soundfield. The system determines a rotation angle for rotating the transmitting soundfield based on a desired location for the sound source. The transmitting soundfield is rotated by the determined angle and the system obtains a listener's soundfield based on the rotated transmitting soundfield. The listener's soundfield is transmitted for rendering to a listener.

Patent Agency Ranking